Displaying 20 results from an estimated 90 matches similar to: "Asterisk.NET authentication problem"
2006 Dec 29
0
PHP to call script
Using the php script below. I am able to enter my number and the number to
call, however I get the following error:
-- AGI Script cid-spoof.agi completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing Wait("OutgoingSpoolFailed",
2005 Jun 22
2
asterisk authentication issue
Hi guys
I am currently getting the following in my log
asterisk1*CLI> Parsing '/etc/asterisk/manager.conf': Found
asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf': Found
asterisk1*CLI> == Connect attempt from '127.0.0.1' unable to authenticate
Can anyone tell me why asterisk would not be able to authenticate it's self?
2006 Jan 27
3
paging agi
Hello Everyone,
I've been playing with an agi script for paging sip phones.
page.agi will take all available sip extensions and assign them to the
global variable PAGE_GROUP. Allowing the phones to be paged from the
dialplan with the new Page cmd. Extensions to be excluded are presented as
arguments to the agi. Each time a page is made this agi refreshes the global
variable. This works with
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Manager Connected msg.
Your input will be much appreciated. I am thinking I have some syntax
2005 May 22
0
*@home 1.0 FWD inbound problems, 2 calls generated
Hi ALL
Have installed asterisk@home 1.0
On FWD DID's, appears that 2 calls are generated to the inbound extention. I
have confirmed this on a number of friends boxes also. Does anyone have a fix
for this ? I set the DID simply to a custom context and it did the same...
Anyone have a way to fix this ?
Here is the output......
-- Accepting AUTHENTICATED call from 65.39.205.121, requested
2005 Feb 02
1
Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients
Hi,
I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
queues.conf :
[prodaja]
music = default
announce = queue-markq
strategy = ringall
context = from-pstn
timeout = 15
retry = 5
maxlen = 0
announce-holdtime = no
announce-frequency = 30
announce-holdtime = yes
monitor-format = gsm|wav|wav49
monitor-join = yes
eventwhencalled = yes
member => Agent/1000
2008 Oct 19
4
Asterisk Problem
After installing a new box and asterisk. i have got these errors
[root at localhost ~]# asterisk
Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
[root at localhost ~]# asterisk -vr
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
I didn't find a folder called asterisk in the directory /var/run
[root at
2009 Dec 29
1
SkyHost is set to expire
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
Hi all,<br>
I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last
version).<br>
Everything was going fine, but yesterday I've got this messege when
I've tried to restart asterisk
2005 Sep 01
1
dialparties.agi is returning no extensions to dial
Hi,
I set up a ring group. I would like for people who select a certain voice
menu option to ring a list of extensions (I have just one extension in there at
the moment) and if it doesn't answer to go to an extension's voice mail. I am
using a version of asterisk from CVS, last updated a couple of weeks ago.
This line in extensions_addtional.conf sends the call to ringgroup 3 if
2008 Jun 28
0
AMI extenstion state
Hi,
I would like to get the status of asterisk extension with my php program.
*My program as follows,*
<html>
<!--<meta http-equiv="refresh" content="1" />-->
<?php
$fp = fsockopen("xxx.xxx.xxx.xxx", 5038, $errno, $errstr, 30);
if (!$fp)
{
echo "$errstr ($errno)<br />\n";
}
else
{
$out = "Action: Login\r\n";
$out
2005 Jul 13
1
Polycoms and paging
I'm looking at deploying some Polycom 501's here, but one thing that still
needs confirmation before I can move forward is global paging.
I figure that I can couple polycom auto-answer
(http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config) with
this script:
http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html
However, that script was posted over a
2010 Feb 02
0
Issue when reloading
Hello list!
I?m having an issue when reloading Asterisk, I?ve had this problem in
Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same
error.
For example, I send a "reload" in Asterisk CLI and this is the output:
isb152*CLI> reload
== Parsing '/etc/asterisk/extconfig.conf': == Found
== Parsing '/etc/asterisk/manager.conf': == Found
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part --------------
asterisk1*CLI> soft hangup Zap/1-1
Requested Hangup on channel 'Zap/1-1'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
== Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
--
2005 Sep 27
1
Extensions go straight to voicemail
Hello,
I have setup a test server with asterisk/AMP and have several 7960's
connected to it. The asterisk server has a public ip and all the
7960's are behind nat'd routers. When I try to call from extension
to extension I get directed straight to voicemail. I do not have any
cards installed and instead direct everything to an Ondo server. I
have been told it's not an AMP
2005 Sep 09
2
AMP 1.10.009 released!
Hello all,
Asterisk Management Portal 1.10.009 has now been released. This
exciting new version has several notable additions (listed below).
The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find
links to the download, install guide, and documentation wiki.
As usual, please use amportal-users mailing list for discussions about
AMP:
2005 Aug 04
1
no ring to callers?
OK, i've got asterisk @ home 1.3 up and running with Broadvoice.
BUT I have one nagging problem to sort out. When you call my BV # the
calling party gets no ring indication, just silence until either I
answer the phone, or the call bounces over to voicemail. below is the
console output when a call is recieved. what am i missing here?
thanks
Bernie
-- Executing
2012 Feb 02
0
Wine release 1.4-rc2
The Wine development release 1.4-rc2 is now available.
What's new in this release (see below for details):
- Bug fixes only, we are in code freeze.
The source is available from the following locations:
http://ibiblio.org/pub/linux/system/emulators/wine/wine-1.4-rc2.tar.bz2
http://prdownloads.sourceforge.net/wine/wine-1.4-rc2.tar.bz2
Binary packages for various distributions will be
2004 Sep 23
10
MFC/R2
Hi all,
I have begun the release of my MFC/R2 protocol software. At
http://www.opencall.org/installing-mfcr2.html there are instructions for
installing what I have released so far. This is the MFC/R2 protocol
software, and a test program. The software to interface Asterisk to the
MFC/R2 code will be released shortly. It used to work, but it hasn't
been touched for a while, and Asterisk
2005 Aug 05
0
Another problem on queues
Hello all,
I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning.
I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically.
If there are no agents, queue timesout and gets derived to another queue that somebody
2009 May 16
1
Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for
a queue up to 15 agents through a PRI line, it was working fine for more than 1
year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs